@abrar71/lib-jitsi-meet 0.0.0

package/dist/esm/modules/statistics/RTPStatsCollector.js

@@ -0,0 +1,601 @@
+import { getLogger } from '@jitsi/logger';
+import { MediaType } from '../../service/RTC/MediaType';
+import { StatisticsEvents } from '../../service/statistics/Events';
+import browser from '../browser';
+import FeatureFlags from '../flags/FeatureFlags';
+import { isValidNumber } from '../util/MathUtil';
+const logger = getLogger('stats:RTPStatsCollector');
+/**
+ * Calculates packet lost percent using the number of lost packets and the
+ * number of all packet.
+ * @param lostPackets the number of lost packets
+ * @param totalPackets the number of all packets.
+ * @returns {number} packet loss percent
+ */
+function calculatePacketLoss(lostPackets, totalPackets) {
+    if (lostPackets > 0 && totalPackets > 0) {
+        return Math.round(lostPackets / totalPackets * 100);
+    }
+    return 0;
+}
+/**
+ * Holds "statistics" for a single SSRC.
+ */
+class SsrcStats {
+    /**
+     * Constructor.
+     */
+    constructor() {
+        this.loss = {};
+        this.bitrate = {
+            download: 0,
+            upload: 0
+        };
+        this.resolution = {};
+        this.framerate = 0;
+        this.codec = '';
+    }
+    /**
+     * Sets the "loss" object.
+     * @param loss the value to set.
+     */
+    setLoss(loss) {
+        this.loss = loss || {};
+    }
+    /**
+     * Sets resolution that belong to the ssrc represented by this instance.
+     * @param resolution new resolution value to be set.
+     */
+    setResolution(resolution) {
+        this.resolution = resolution || {};
+    }
+    /**
+     * Adds the "download" and "upload" fields from the "bitrate" parameter to
+     * the respective fields of the "bitrate" field of this object.
+     * @param bitrate an object holding the values to add.
+     */
+    addBitrate(bitrate) {
+        this.bitrate.download += bitrate.download;
+        this.bitrate.upload += bitrate.upload;
+    }
+    /**
+     * Resets the bit rate for given <tt>ssrc</tt> that belong to the peer
+     * represented by this instance.
+     */
+    resetBitrate() {
+        this.bitrate.download = 0;
+        this.bitrate.upload = 0;
+    }
+    /**
+     * Sets the "framerate".
+     * @param framerate the value to set.
+     */
+    setFramerate(framerate) {
+        this.framerate = framerate || 0;
+    }
+    /**
+     * Sets the codec.
+     * @param codec the value to set.
+     */
+    setCodec(codec) {
+        this.codec = codec || '';
+    }
+    /**
+     * Sets the encode stats.
+     * @param encodeStats the value to set.
+     */
+    setEncodeStats(encodeStats) {
+        this.encodeStats = encodeStats || {};
+    }
+}
+/**
+ * Conference statistics.
+ */
+class ConferenceStats {
+    /**
+     * Constructor.
+     */
+    constructor() {
+        /**
+         * The bandwidth
+         * @type {{}}
+         */
+        this.bandwidth = {};
+        /**
+         * The bit rate
+         * @type {{}}
+         */
+        this.bitrate = {};
+        /**
+         * The packet loss rate
+         * @type {{}}
+         */
+        this.packetLoss = null;
+        /**
+         * Array with the transport information.
+         * @type {Array}
+         */
+        this.transport = [];
+    }
+}
+/* eslint-disable max-params */
+/**
+ * <tt>StatsCollector</tt> registers for stats updates of given
+ * <tt>peerconnection</tt> in given <tt>interval</tt>. On each update particular
+ * stats are extracted and put in {@link SsrcStats} objects. Once the processing
+ * is done <tt>audioLevelsUpdateCallback</tt> is called with <tt>this</tt>
+ * instance as an event source.
+ */
+export default class StatsCollector {
+    /**
+     * Creates new <tt>StatsCollector</tt> instance.
+     * @param peerconnection WebRTC PeerConnection object.
+     * @param audioLevelsInterval
+     * @param statsInterval stats refresh interval given in ms.
+     * @param eventEmitter
+     * @constructor
+     */
+    constructor(peerconnection, audioLevelsInterval, statsInterval, eventEmitter) {
+        this.peerconnection = peerconnection;
+        this.currentStatsReport = null;
+        this.previousStatsReport = null;
+        this.audioLevelsIntervalId = null;
+        this.eventEmitter = eventEmitter;
+        this.conferenceStats = new ConferenceStats();
+        // Updates stats interval
+        this.audioLevelsIntervalMilis = audioLevelsInterval;
+        this.speakerList = [];
+        this.statsIntervalId = null;
+        this.statsIntervalMilis = statsInterval;
+        /**
+         * Maps SSRC numbers to {@link SsrcStats}.
+         * @type {Map<number,SsrcStats}
+         */
+        this.ssrc2stats = new Map();
+    }
+    /**
+     * Set the list of the remote speakers for which audio levels are to be calculated.
+     *
+     * @param {Array<string>} speakerList - Endpoint ids.
+     * @returns {void}
+     */
+    setSpeakerList(speakerList) {
+        this.speakerList = speakerList;
+    }
+    /**
+     * Stops stats updates.
+     */
+    stop() {
+        if (this.audioLevelsIntervalId) {
+            clearInterval(this.audioLevelsIntervalId);
+            this.audioLevelsIntervalId = null;
+        }
+        if (this.statsIntervalId) {
+            clearInterval(this.statsIntervalId);
+            this.statsIntervalId = null;
+        }
+    }
+    /**
+     * Callback passed to <tt>getStats</tt> method.
+     * @param error an error that occurred on <tt>getStats</tt> call.
+     */
+    errorCallback(error) {
+        logger.error('Get stats error', error);
+        this.stop();
+    }
+    /**
+     * Starts stats updates.
+     */
+    start(startAudioLevelStats) {
+        if (startAudioLevelStats && browser.supportsReceiverStats()) {
+            this.audioLevelsIntervalId = setInterval(() => {
+                const audioLevels = this.peerconnection.getAudioLevels(this.speakerList);
+                for (const ssrc in audioLevels) {
+                    if (audioLevels.hasOwnProperty(ssrc)) {
+                        // Use a scaling factor of 2.5 to report the same audio levels that getStats reports.
+                        const audioLevel = audioLevels[ssrc] * 2.5;
+                        this.eventEmitter.emit(StatisticsEvents.AUDIO_LEVEL, this.peerconnection, Number.parseInt(ssrc, 10), audioLevel, false /* isLocal */);
+                    }
+                }
+            }, this.audioLevelsIntervalMilis);
+        }
+        const processStats = () => {
+            // Interval updates
+            this.peerconnection.getStats()
+                .then(report => {
+                this.currentStatsReport = typeof report?.result === 'function'
+                    ? report.result()
+                    : report;
+                try {
+                    this.processStatsReport();
+                }
+                catch (error) {
+                    logger.error('Processing of RTP stats failed:', error);
+                }
+                this.previousStatsReport = this.currentStatsReport;
+            })
+                .catch(error => this.errorCallback(error));
+        };
+        processStats();
+        this.statsIntervalId = setInterval(processStats, this.statsIntervalMilis);
+    }
+    /**
+     * Process and emit statistics report.
+     * @private
+     */
+    _processAndEmitReport() {
+        // process stats
+        const totalPackets = {
+            download: 0,
+            upload: 0
+        };
+        const lostPackets = {
+            download: 0,
+            upload: 0
+        };
+        let bitrateDownload = 0;
+        let bitrateUpload = 0;
+        const resolutions = {};
+        const framerates = {};
+        const codecs = {};
+        let audioBitrateDownload = 0;
+        let audioBitrateUpload = 0;
+        let videoBitrateDownload = 0;
+        let videoBitrateUpload = 0;
+        for (const [ssrc, ssrcStats] of this.ssrc2stats) {
+            // process packet loss stats
+            const loss = ssrcStats.loss;
+            const type = loss.isDownloadStream ? 'download' : 'upload';
+            totalPackets[type] += loss.packetsTotal;
+            lostPackets[type] += loss.packetsLost;
+            const ssrcBitrateDownload = ssrcStats.bitrate.download;
+            const ssrcBitrateUpload = ssrcStats.bitrate.upload;
+            // process bitrate stats
+            bitrateDownload += ssrcBitrateDownload;
+            bitrateUpload += ssrcBitrateUpload;
+            ssrcStats.resetBitrate();
+            // collect resolutions and framerates
+            const track = this.peerconnection.getTrackBySSRC(ssrc);
+            if (!track) {
+                continue; // eslint-disable-line no-continue
+            }
+            let audioCodec;
+            let videoCodec;
+            if (track.isAudioTrack()) {
+                audioBitrateDownload += ssrcBitrateDownload;
+                audioBitrateUpload += ssrcBitrateUpload;
+                audioCodec = ssrcStats.codec;
+            }
+            else {
+                videoBitrateDownload += ssrcBitrateDownload;
+                videoBitrateUpload += ssrcBitrateUpload;
+                videoCodec = ssrcStats.codec;
+            }
+            const participantId = track.getParticipantId();
+            if (!participantId) {
+                // All tracks in ssrc-rewriting mode need not have a participant associated with it.
+                if (!FeatureFlags.isSsrcRewritingSupported()) {
+                    logger.error(`No participant ID returned by ${track}`);
+                }
+                continue;
+            }
+            const userCodecs = codecs[participantId] ?? {};
+            userCodecs[ssrc] = {
+                audio: audioCodec,
+                video: videoCodec
+            };
+            codecs[participantId] = userCodecs;
+            const { resolution } = ssrcStats;
+            if (!track.isVideoTrack()
+                || !isValidNumber(resolution?.height)
+                || !isValidNumber(resolution?.width)
+                || resolution.height === -1
+                || resolution.width === -1) {
+                continue;
+            }
+            const userResolutions = resolutions[participantId] || {};
+            // If simulcast (VP8) is used, there will be 3 "outbound-rtp" streams with different resolutions and 3
+            // different SSRCs. Based on the requested resolution and the current cpu and available bandwidth
+            // values, some of the streams might get suspended. Therefore the actual send resolution needs to be
+            // calculated based on the outbound-rtp streams that are currently active for the simulcast case.
+            // However for the SVC case, there will be only 1 "outbound-rtp" stream which will have the correct
+            // send resolution width and height.
+            if (track.isLocal() && !browser.supportsTrackBasedStats() && this.peerconnection.doesTrueSimulcast(track)) {
+                const localSsrcs = this.peerconnection.getLocalVideoSSRCs(track);
+                for (const localSsrc of localSsrcs) {
+                    const ssrcResolution = this.ssrc2stats.get(localSsrc)?.resolution;
+                    // The code processes resolution stats only for 'outbound-rtp' streams that are currently active.
+                    if (ssrcResolution?.height && ssrcResolution?.width) {
+                        resolution.height = Math.max(resolution.height, ssrcResolution.height);
+                        resolution.width = Math.max(resolution.width, ssrcResolution.width);
+                    }
+                }
+            }
+            userResolutions[ssrc] = resolution;
+            resolutions[participantId] = userResolutions;
+            if (ssrcStats.framerate > 0) {
+                const userFramerates = framerates[participantId] || {};
+                userFramerates[ssrc] = ssrcStats.framerate;
+                framerates[participantId] = userFramerates;
+            }
+        }
+        this.conferenceStats.bitrate = {
+            download: bitrateDownload,
+            upload: bitrateUpload
+        };
+        this.conferenceStats.bitrate.audio = {
+            download: audioBitrateDownload,
+            upload: audioBitrateUpload
+        };
+        this.conferenceStats.bitrate.video = {
+            download: videoBitrateDownload,
+            upload: videoBitrateUpload
+        };
+        this.conferenceStats.packetLoss = {
+            download: calculatePacketLoss(lostPackets.download, totalPackets.download),
+            total: calculatePacketLoss(lostPackets.download + lostPackets.upload, totalPackets.download + totalPackets.upload),
+            upload: calculatePacketLoss(lostPackets.upload, totalPackets.upload)
+        };
+        this.eventEmitter.emit(StatisticsEvents.CONNECTION_STATS, this.peerconnection, {
+            bandwidth: this.conferenceStats.bandwidth,
+            bitrate: this.conferenceStats.bitrate,
+            codec: codecs,
+            framerate: framerates,
+            packetLoss: this.conferenceStats.packetLoss,
+            resolution: resolutions,
+            transport: this.conferenceStats.transport
+        });
+        this.conferenceStats.transport = [];
+    }
+    /**
+     * Converts the value to a non-negative number.
+     * If the value is either invalid or negative then 0 will be returned.
+     * @param {*} v
+     * @return {number}
+     * @private
+     */
+    getNonNegativeValue(v) {
+        let value = v;
+        if (typeof value !== 'number') {
+            value = Number(value);
+        }
+        if (!isValidNumber(value)) {
+            return 0;
+        }
+        return Math.max(0, value);
+    }
+    /**
+     * Calculates bitrate between before and now using a supplied field name and its
+     * value in the stats.
+     * @param {RTCInboundRtpStreamStats|RTCSentRtpStreamStats} now the current stats
+     * @param {RTCInboundRtpStreamStats|RTCSentRtpStreamStats} before the
+     * previous stats.
+     * @param fieldName the field to use for calculations.
+     * @return {number} the calculated bitrate between now and before.
+     * @private
+     */
+    _calculateBitrate(now, before, fieldName) {
+        const bytesNow = this.getNonNegativeValue(now[fieldName]);
+        const bytesBefore = this.getNonNegativeValue(before[fieldName]);
+        const bytesProcessed = Math.max(0, bytesNow - bytesBefore);
+        const timeMs = now.timestamp - before.timestamp;
+        let bitrateKbps = 0;
+        if (timeMs > 0) {
+            // TODO is there any reason to round here?
+            bitrateKbps = Math.round((bytesProcessed * 8) / timeMs);
+        }
+        return bitrateKbps;
+    }
+    /**
+     * Calculates the frames per second rate between before and now using a supplied field name and its value in stats.
+     * @param {RTCOutboundRtpStreamStats|RTCSentRtpStreamStats} now the current stats
+     * @param {RTCOutboundRtpStreamStats|RTCSentRtpStreamStats} before the previous stats
+     * @param {string} fieldName the field to use for calculations.
+     * @returns {number} the calculated frame rate between now and before.
+     */
+    _calculateFps(now, before, fieldName) {
+        const timeMs = now.timestamp - before.timestamp;
+        let frameRate = 0;
+        if (timeMs > 0 && now[fieldName]) {
+            const numberOfFramesSinceBefore = now[fieldName] - before[fieldName];
+            frameRate = (numberOfFramesSinceBefore / timeMs) * 1000;
+        }
+        return frameRate;
+    }
+    /**
+     * Stats processing for spec-compliant RTCPeerConnection#getStats.
+     */
+    processStatsReport() {
+        const byteSentStats = {};
+        const encodedTimeStatsPerSsrc = new Map();
+        this.currentStatsReport.forEach(now => {
+            const before = this.previousStatsReport ? this.previousStatsReport.get(now.id) : null;
+            // RTCIceCandidatePairStats - https://w3c.github.io/webrtc-stats/#candidatepair-dict*
+            if (now.type === 'candidate-pair' && now.nominated && now.state === 'succeeded') {
+                const availableIncomingBitrate = now.availableIncomingBitrate;
+                const availableOutgoingBitrate = now.availableOutgoingBitrate;
+                if (availableIncomingBitrate || availableOutgoingBitrate) {
+                    this.conferenceStats.bandwidth = {
+                        'download': Math.round(availableIncomingBitrate / 1000),
+                        'upload': Math.round(availableOutgoingBitrate / 1000)
+                    };
+                }
+                const remoteUsedCandidate = this.currentStatsReport.get(now.remoteCandidateId);
+                const localUsedCandidate = this.currentStatsReport.get(now.localCandidateId);
+                // RTCIceCandidateStats
+                // https://w3c.github.io/webrtc-stats/#icecandidate-dict*
+                if (remoteUsedCandidate && localUsedCandidate) {
+                    const remoteIpAddress = browser.isChromiumBased()
+                        ? remoteUsedCandidate.ip
+                        : remoteUsedCandidate.address;
+                    const remotePort = remoteUsedCandidate.port;
+                    const ip = `${remoteIpAddress}:${remotePort}`;
+                    const localIpAddress = browser.isChromiumBased()
+                        ? localUsedCandidate.ip
+                        : localUsedCandidate.address;
+                    const localPort = localUsedCandidate.port;
+                    const localip = `${localIpAddress}:${localPort}`;
+                    const type = remoteUsedCandidate.protocol;
+                    // Save the address unless it has been saved already.
+                    const conferenceStatsTransport = this.conferenceStats.transport;
+                    if (!conferenceStatsTransport.some(t => t.ip === ip
+                        && t.type === type
+                        && t.localip === localip)) {
+                        conferenceStatsTransport.push({
+                            ip,
+                            localCandidateType: localUsedCandidate.candidateType,
+                            localip,
+                            networkType: localUsedCandidate.networkType,
+                            p2p: this.peerconnection.isP2P,
+                            remoteCandidateType: remoteUsedCandidate.candidateType,
+                            rtt: now.currentRoundTripTime * 1000,
+                            type
+                        });
+                    }
+                }
+                // RTCReceivedRtpStreamStats
+                // https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
+                // RTCSentRtpStreamStats
+                // https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*
+            }
+            else if (now.type === 'inbound-rtp' || now.type === 'outbound-rtp') {
+                const ssrc = this.getNonNegativeValue(now.ssrc);
+                if (!ssrc) {
+                    return;
+                }
+                let ssrcStats = this.ssrc2stats.get(ssrc);
+                if (!ssrcStats) {
+                    ssrcStats = new SsrcStats();
+                    this.ssrc2stats.set(ssrc, ssrcStats);
+                }
+                let isDownloadStream = true;
+                let key = 'packetsReceived';
+                if (now.type === 'outbound-rtp') {
+                    isDownloadStream = false;
+                    key = 'packetsSent';
+                }
+                let packetsNow = now[key];
+                if (!packetsNow || packetsNow < 0) {
+                    packetsNow = 0;
+                }
+                if (before) {
+                    const packetsBefore = this.getNonNegativeValue(before[key]);
+                    const packetsDiff = Math.max(0, packetsNow - packetsBefore);
+                    const packetsLostNow = this.getNonNegativeValue(now.packetsLost);
+                    const packetsLostBefore = this.getNonNegativeValue(before.packetsLost);
+                    const packetsLostDiff = Math.max(0, packetsLostNow - packetsLostBefore);
+                    ssrcStats.setLoss({
+                        isDownloadStream,
+                        packetsLost: packetsLostDiff,
+                        packetsTotal: packetsDiff + packetsLostDiff
+                    });
+                }
+                let resolution;
+                // Process the stats for 'inbound-rtp' streams always and 'outbound-rtp' only if the browser is
+                // Chromium based and version 112 and later since 'track' based stats are no longer available there
+                // for calculating send resolution and frame rate.
+                if (typeof now.frameHeight !== 'undefined' && typeof now.frameWidth !== 'undefined') {
+                    // Assume the stream is active if the field is missing in the stats(Firefox)
+                    const isStreamActive = now.active ?? true;
+                    if (now.type === 'inbound-rtp' || (!browser.supportsTrackBasedStats() && isStreamActive)) {
+                        resolution = {
+                            height: now.frameHeight,
+                            width: now.frameWidth
+                        };
+                    }
+                }
+                ssrcStats.setResolution(resolution);
+                let frameRate = now.framesPerSecond;
+                if (!frameRate && before) {
+                    frameRate = this._calculateFps(now, before, 'framesSent');
+                }
+                ssrcStats.setFramerate(Math.round(frameRate || 0));
+                if (now.type === 'inbound-rtp' && before) {
+                    ssrcStats.addBitrate({
+                        'download': this._calculateBitrate(now, before, 'bytesReceived'),
+                        'upload': 0
+                    });
+                }
+                else if (before) {
+                    byteSentStats[ssrc] = this.getNonNegativeValue(now.bytesSent);
+                    ssrcStats.addBitrate({
+                        'download': 0,
+                        'upload': this._calculateBitrate(now, before, 'bytesSent')
+                    });
+                }
+                const codec = this.currentStatsReport.get(now.codecId);
+                if (codec) {
+                    /**
+                     * The mime type has the following form: video/VP8 or audio/ISAC, so we what to keep just the type
+                     * after the '/', audio and video keys will be added on the processing side.
+                     */
+                    const codecShortType = codec.mimeType.split('/')[1];
+                    codecShortType && ssrcStats.setCodec(codecShortType);
+                    // Calculate the encodeTime stat for outbound video streams.
+                    const track = this.peerconnection.getTrackBySSRC(ssrc);
+                    if (now.type === 'outbound-rtp'
+                        && now.active
+                        && track?.isVideoTrack()
+                        && before?.totalEncodeTime
+                        && before?.framesEncoded
+                        && now.frameHeight
+                        && now.frameWidth) {
+                        const encodeTimeDelta = now.totalEncodeTime - before.totalEncodeTime;
+                        const framesEncodedDelta = now.framesEncoded - before.framesEncoded;
+                        const encodeTimePerFrameInMs = 1000 * encodeTimeDelta / framesEncodedDelta;
+                        const encodeTimeStats = {
+                            codec: codecShortType,
+                            encodeTime: encodeTimePerFrameInMs,
+                            qualityLimitationReason: now.qualityLimitationReason,
+                            resolution,
+                            timestamp: now.timestamp
+                        };
+                        encodedTimeStatsPerSsrc.set(ssrc, encodeTimeStats);
+                        ssrcStats.setEncodeStats(encodedTimeStatsPerSsrc);
+                    }
+                }
+                // Continue to use the 'track' based stats for Firefox and Safari and older versions of Chromium.
+            }
+            else if (browser.supportsTrackBasedStats()
+                && now.type === 'track'
+                && now.kind === MediaType.VIDEO
+                && !now.remoteSource) {
+                const resolution = {
+                    height: now.frameHeight,
+                    width: now.frameWidth
+                };
+                const localVideoTracks = this.peerconnection.getLocalTracks(MediaType.VIDEO);
+                if (!localVideoTracks?.length) {
+                    return;
+                }
+                const ssrc = this.peerconnection.getSsrcByTrackId(now.trackIdentifier);
+                if (!ssrc) {
+                    return;
+                }
+                let ssrcStats = this.ssrc2stats.get(ssrc);
+                if (!ssrcStats) {
+                    ssrcStats = new SsrcStats();
+                    this.ssrc2stats.set(ssrc, ssrcStats);
+                }
+                if (resolution.height && resolution.width) {
+                    ssrcStats.setResolution(resolution);
+                }
+                // Calculate the frame rate. 'framesSent' is the total aggregate value for all the simulcast streams.
+                // Therefore, it needs to be divided by the total number of active simulcast streams.
+                let frameRate = now.framesPerSecond;
+                if (!frameRate && before) {
+                    frameRate = this._calculateFps(now, before, 'framesSent');
+                }
+                ssrcStats.setFramerate(frameRate);
+            }
+        });
+        if (Object.keys(byteSentStats).length) {
+            this.eventEmitter.emit(StatisticsEvents.BYTE_SENT_STATS, this.peerconnection, byteSentStats);
+        }
+        if (encodedTimeStatsPerSsrc.size) {
+            this.eventEmitter.emit(StatisticsEvents.ENCODE_TIME_STATS, this.peerconnection, encodedTimeStatsPerSsrc);
+        }
+        this._processAndEmitReport();
+    }
+}
+//# sourceMappingURL=RTPStatsCollector.js.map

package/dist/esm/modules/statistics/RTPStatsCollector.js.map

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